EP1295444B1 - Procede de reduction de la distorsion dans la transmission de la voix sur des reseaux de donnees - Google Patents

Procede de reduction de la distorsion dans la transmission de la voix sur des reseaux de donnees Download PDF

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Publication number
EP1295444B1
EP1295444B1 EP01945473.5A EP01945473A EP1295444B1 EP 1295444 B1 EP1295444 B1 EP 1295444B1 EP 01945473 A EP01945473 A EP 01945473A EP 1295444 B1 EP1295444 B1 EP 1295444B1
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EP
European Patent Office
Prior art keywords
voice data
computer telephony
digital voice
communications network
packets
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Expired - Lifetime
Application number
EP01945473.5A
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German (de)
English (en)
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EP1295444A1 (fr
Inventor
Richard John Buchan Reynolds
Phillip Gray
Michael Peter Hollier
Antony William Rix
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British Telecommunications PLC
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British Telecommunications PLC
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Priority to EP01945473.5A priority Critical patent/EP1295444B1/fr
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Publication of EP1295444B1 publication Critical patent/EP1295444B1/fr
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/80—Responding to QoS
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066—Session management
    • H04L65/1101—Session protocols
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M7/00—Arrangements for interconnection between switching centres
    • H04M7/006—Networks other than PSTN/ISDN providing telephone service, e.g. Voice over Internet Protocol (VoIP), including next generation networks with a packet-switched transport layer

Definitions

  • This invention relates to a signal processing method for use with computer based telephony.
  • the computers are transmitting real-time data, such as telephony or audio-visual signals for example, then the reception of out of sequence packets and the failure to receive some packets in time, or not at all, may cause significant problems with the playback of the real-time data.
  • real-time data such as telephony or audio-visual signals for example
  • the invention provides a method of transmitting computer telephony packets over a communications network, the method comprising the steps of:
  • Step (iii) may comprise applying a band pass filter to the plurality of voice data packets and in particular this step may comprise filtering frequency components less than substantially 300 Hz from the plurality of voice data packets and/or filtering frequency components greater than substantially 3400 Hz from the plurality of voice data packets.
  • Step (iii) may comprise attenuating one or more frequency components from the plurality of voice data packets and/or amplifying one or more frequency components from the plurality of voice data packets.
  • Figure 1 shows a functional block diagram of a computer telephony terminal.
  • the computer telephony terminal is implemented as a computer.
  • the computer includes a central processing unit, memory and storage devices and various hardware devices such as a display unit and a keyboard.
  • the computer has an operating system (indicated by reference numeral 100 in Figure 1 ) stored in the memory and storage devices.
  • Software applications such as a word processor application and an Internet browser may also be stored in the memory and storage devices and run under the operating system.
  • the computer includes a software application and hardware to provide it with the functionality of a computer telephony terminal.
  • the computer telephony terminal includes a computer telephony (CT) software application 10, which is a process that is run under the computer operating system 100.
  • CT computer telephony
  • the CT application is used to establish a communication link with a CT application which is being run on a second computer, which is connected to the first computer via a communications network. It is also possible for the CT application to communicate with a standard telephone which is connected to the PSTN (public switched telephony network) if there is a suitable gateway from the communications network to the PSTN.
  • PSTN public switched telephony network
  • the CT application may communicate with a so-called 'internet phone' which has the appearance and functionality of a conventional telephone but instead of connecting to, and communicating via, the PSTN such 'internet phones' are connected to the communication network and provide the same functionality as a computer running a CT application but at much reduced cost and complexity (see " Voice-Data Convergence and the Corporate Voice-Over-IP Trial", A Catchpole, IBTE Journal, volume 17, part 4, January 1999, pages 218-224 ).
  • the CT application 10 communicates with a CT application in an other computer, or internet phone, by sending and receiving data transmission packets, the data packets containing short segments of speech.
  • the CT application 10 communicates with the computer's soundcard 20, which is connected to a microphone 21 and a loudspeaker(s) 22 (the microphone and loudspeaker(s) may be integrated and take the form of a headset or a conventional telephony handset).
  • the user's voice signal is captured by the microphone 21 and then the soundcard converts the analogue signal supplied by the microphone into digital voice data.
  • This digital voice data is then sent to the CT application, which may perform some form of data modulation or compression in order to increase the efficiency of the data transmission across the network before the digital voice data is arranged into packets of a suitable size (either in terms of the number of bytes of the package or in terms of the length of speech that is represented by the digital voice data).
  • These voice data packets are sent to the communications interface 30 such that the voice data therein may be transmitted to the destination computer.
  • the communications interface 30 may be a modem for dial-up access to a packet-based network or it may be a network card for direct access to a packet-based network.
  • the communications interface places the voice data packets within the payload of the data transmission packets used by the communications interface, for example TCP (Transmission Control Protocol), UDP (User Datagram Protocol), etc.
  • TCP Transmission Control Protocol
  • UDP User Datagram Protocol
  • the data transmission packets are then routed across the communications network to the destination computer, where the analogue speech signal is recovered in a manner described below.
  • the communications interface 30 receives data transmission packets that have been sent to the computer and differentiates the data transmission packets by their content, for example part of a stream of computer telephony packets, a WWW download, an email message, etc.
  • the voice data packets are extracted from the payload of the data transmission packets and are then sent to the CT application 10.
  • the CT application processes the voice data packets, ordering them in the correct sequence and demodulating (and/or decompressing) the data as necessary so that a stream of digital voice data can be sent to the soundcard 20, which converts the data from a digital signal to an analogue signal, which can then be played back to the user through the loudspeaker(s) 22.
  • the soundcard provides functionality using both hardware 28, which interfaces with the external hardware (loudspeakers, microphone, etc.) and the internal bus of the computer, and software drivers 25 stored in a memory device, which may be mounted on the soundcard, which allow the operating system and the applications running on the computer to access functions that cause the hardware to respond accordingly.
  • Personal computers typically are very versatile, enabling the user to add functionality to the PC by adding hardware and software, e.g. to enable DVD movie playback, receiving and playing back television and radio signals, reading CD-ROMs, etc.
  • This wide variety of peripherals means that there is a very large number of different permutations of soundcard, loudspeakers, headphones and microphones that a user may have in a PC.
  • these devices can be set up in accordance with the user's preferences (either through selecting options within software programs or through physical placement of objects) it is clear that very few PCs will have identical audio properties. Due to the multi-functional nature of a personal computer, one audio set-up will be used for a number of different uses (e.g.
  • the inventors have realised that the multi-functional set-up of a personal computer severely compromises the ability of the computer to provide computer telephony functionality, as the codecs (coder/decoder) commonly used in CT applications, such as GSM, ADPCM, etc. are designed to work with signals having the frequency and level that are frequently experienced with conventional telephony terminals.
  • codecs coder/decoder
  • GSM Global System for Mobile Communications
  • ADPCM ADPCM
  • FIG. 2 shows a schematic representation of a computer telephony terminal incorporating the present invention.
  • the CT terminal shown in Figure 2 is generally similar to the CT terminal of Figure 1 but modified as described below.
  • Signal processing application 50 is a software process running under the computer operating system in parallel with the CT application 10 and any other software applications.
  • the soundcard software drivers 25 shown in Figure 1 have been replaced with modified soundcard drivers 35. These modified drivers have all the functionality required to operate under the operating system 100 (as do the soundcard drivers 25 shown in Figure 1 ) plus additional functionality in order to interface with the signal processing application.
  • the CT application additionally comprises one or more codecs for use when transforming the received voice signal into digital voice data.
  • the codec(s) enable the digital voice data to be compressed to varying levels for example a PCM codec (as used in PSTNs) will give a 64kbit/s output and a GSM codec gives a 9.6 kbit/s output.
  • PCM codec as used in PSTNs
  • GSM codec gives a 9.6 kbit/s output.
  • DPCM and ADPCM can be also be used to provide digital voice data streams, typically providing a signal having an equivalent quality to that of PCM but requiring less bandwidth (e.g. 32kbit/s or 16 kbit/s).
  • analogue voice signals are received by the soundcard from the microphone.
  • the soundcard transforms these analogue signals into the digital domain and sends the digital voice data to the signal processing application 50.
  • the signal processing application analyses the received digital voice data in order to transform the digital voice data so that it can be presented to the codec in the CT application in a more suitable format.
  • Each of the codecs are optimised to code signals that normally have frequency components from the extremes of human speech removed or attenuated, for example the components below 300 Hz and above 3.4 kHz will normally be filtered.
  • the inventors have observed that sending signals with significant low frequency components to codecs has led to a very muffled sound when the output of the codec is decoded. It has been postulated that the energy of the low frequency sound components overwhelms the codec, preventing it from coding the input signals correctly. As the lower frequencies in this case appear to cause the most significant problems, it may only be necessary to filter out the lower frequencies, although there may be some systems for which only filtering out the higher frequency components is applicable.
  • the amplitude of the digital voice data may be processed before the data is presented to the CT application, in order to bring it within an expected maximum threshold value. If an analysis of the digital voice data indicates that the audio signals contained a skew, towards either the lower or the higher frequency end of the signal spectrum, or if a frequency band is more pronounced than would be expected, then the digital voice data can be manipulated to remove the skew to the spectrum or to 'flatten out' any unexpected features.
  • a frequency template For each codec a frequency template can be generated which defines the upper and lower frequency limits, roll-on and roll-off gradients, maximum amplitude values, etc. The exact frequency cut-off point and the amount of filtering applied for the filters can be varied to suit the details of a particular coded and/or transmission system.
  • Such templates can be generated by feeding speech patterns and signals through a codec and assessing the quality of the codec output to determine which frequency components, and at which amplitudes, cause distortion and coding errors.
  • the signal processing application compares the frequency spectrum of the received digital voice data with the frequency template associated with the codec in use and changes the voice data frequency spectrum to be as similar as possible to the frequency template. If the signal processing application is not able to call upon sufficient machine resources to compare every frequency component and then make all the necessary changes in real time (i.e. without introducing an excessive delay into the system) then the most significant frequency components should be prioritised and the comparison (and any necessary) changes should be made for the priority frequency components. Once the digital voice data has been processed in this manner then it is passed to the CT application for further processing prior to CT packets being transmitted.
  • the analysis of the voice data may be performed on a 'one-off' basis when a VoIP call is set up, but it is preferred that such analysis is performed on a frequent basis or continually in order to accommodate any changes in the user setup (i.e. the user moving further from a fixed microphone, changes in background noise, etc.).
  • EP-B-0 647 375 EP-B-0 705 501 , EP-B-0 776 566 & EP-B-0 776 567 .
  • EP-B-0 705 501 EP-B-0 776 566 & EP-B-0 776 567 .
  • the digital voice data can be presented to the CT application in a more suitable form for being coded.
  • Codecs in use in PSTNs have been designed to receive signals that have been constrained in the frequency domain (i.e. low and high frequency components are filtered out from the audio signals) and telephone handsets have been designed in the light of research into human behaviour, resulting in audio signal levels that tend to fall within known thresholds.
  • the signal processing application 50 may also be used to 'clean up' incoming signals so that a better quality of audio signal is presented to the user (i.e. filtering out noise where possible).
  • the processing application may measure the audio signal played back to the user via the microphone used in the CT application. This information may be used to attenuate or boost certain frequency components of the signal that is played back to the user, compensating for the environment in which the user is in or the quality of the users playback equipment.
  • test functionality can be incorporated into a set top box, games console, personal digital assistant (PDA) that provide computer telephony functionality or mobile terminals and handsets that tend to be used in noisy environments or use low-bandwidth and/or low quality communications links.
  • PDA personal digital assistant
  • the invention may be implemented using solely software, solely hardware or a combination of software and hardware.

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  • Engineering & Computer Science (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • Multimedia (AREA)
  • Business, Economics & Management (AREA)
  • General Business, Economics & Management (AREA)
  • Telephonic Communication Services (AREA)
  • Data Exchanges In Wide-Area Networks (AREA)

Claims (7)

  1. Procédé de transmission de paquets de téléphonie informatique sur un réseau de communication, le procédé comprenant les étapes ci-dessous consistant à :
    (a) générer une pluralité de paquets de données vocales en compressant des données vocales numériques au moyen d'un codeur, le codeur étant sélectionné parmi une pluralité de codeurs ;
    (b) traiter ladite pluralité de paquets de données vocales en vue d'obtenir une pluralité de paquets de téléphonie informatique ;
    (c) transmettre ladite pluralité de paquets de téléphonie informatique sur ledit réseau de communication ;
    le procédé étant caractérisé par les étapes supplémentaires ci-dessous consistant à :
    (i) analyser le spectre de fréquences des données vocales numériques ;
    (ii) comparer les spectres de fréquences analysés à un modèle de spectres de fréquences ; et
    (iii) modifier une ou plusieurs composantes de fréquence des spectres de fréquences analysés en vue de réduire la différence entre lesdits spectres de fréquences analysés et les modèle de spectres de fréquences,
    avant de compresser la pluralité de paquets de données vocales au cours de l'étape (a), dans lequel le modèle est associé au codeur sélectionné.
  2. Procédé de transmission de paquets de téléphonie informatique sur un réseau de communication selon la revendication 1, dans lequel l'étape (iii) comprend l'application d'un filtre passe-bande aux données vocales numériques.
  3. Procédé de transmission de paquets de téléphonie informatique sur un réseau de communication selon la revendication 1 ou 2, dans lequel l'étape (iii) comprend le filtrage de composantes de fréquence sensiblement inférieures à 300 Hz à partir des données vocales numériques.
  4. Procédé de transmission de paquets de téléphonie informatique sur un réseau de communication selon la revendication 1 ou 2, dans lequel l'étape (iii) comprend le filtrage de composantes de fréquence sensiblement supérieures à 3 400 Hz à partir des données vocales numériques.
  5. Procédé de transmission de paquets de téléphonie informatique sur un réseau de communication selon la revendication 1, dans lequel l'étape (iii) comprend l'atténuation d'une ou plusieurs composantes de fréquence à partir des données vocales numériques.
  6. Procédé de transmission de paquets de téléphonie informatique sur un réseau de communication selon la revendication 1 ou 5, dans lequel l'étape (iii) comprend l'amplification d'une ou plusieurs composantes de fréquence à partir des données vocales numériques.
  7. Appareil destiné à mettre en oeuvre la totalité des étapes de l'un quelconque des procédés selon les revendications 1 à 6, l'appareil comprenant une application de téléphonie informatique (10), une application de traitement de signal (50), un moyen d'enregistrement audio (21) et un moyen de génération audio (22).
EP01945473.5A 2000-06-26 2001-06-26 Procede de reduction de la distorsion dans la transmission de la voix sur des reseaux de donnees Expired - Lifetime EP1295444B1 (fr)

Priority Applications (1)

Application Number Priority Date Filing Date Title
EP01945473.5A EP1295444B1 (fr) 2000-06-26 2001-06-26 Procede de reduction de la distorsion dans la transmission de la voix sur des reseaux de donnees

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Application Number Priority Date Filing Date Title
EP00305377 2000-06-26
EP00305377A EP1168734A1 (fr) 2000-06-26 2000-06-26 Procédé pour réduir la distorsion d'une transmission de voix par un réseau de données
EP01945473.5A EP1295444B1 (fr) 2000-06-26 2001-06-26 Procede de reduction de la distorsion dans la transmission de la voix sur des reseaux de donnees
PCT/GB2001/002848 WO2002001824A1 (fr) 2000-06-26 2001-06-26 Procede de reduction de la distorsion dans la transmission de la voix sur des reseaux de donnees

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EP01945473.5A Expired - Lifetime EP1295444B1 (fr) 2000-06-26 2001-06-26 Procede de reduction de la distorsion dans la transmission de la voix sur des reseaux de donnees

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EP1519628A3 (fr) 2003-09-29 2009-03-04 Siemens Aktiengesellschaft Procédé et appareil pour la restitution d'un signal de sortie de mode stéréo qui est obtenu d'un signal d'entrée de mode mono
JP4389560B2 (ja) * 2003-11-28 2009-12-24 沖電気工業株式会社 リアルタイム通信システムおよびメディアエンド端末
US9641562B2 (en) * 2011-12-29 2017-05-02 Vonage Business Inc. Systems and methods of monitoring call quality
US20150254340A1 (en) * 2014-03-10 2015-09-10 JamKazam, Inc. Capability Scoring Server And Related Methods For Interactive Music Systems

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EP1168734A1 (fr) 2002-01-02
US20030133440A1 (en) 2003-07-17
US7567548B2 (en) 2009-07-28
EP1295444A1 (fr) 2003-03-26
WO2002001824A1 (fr) 2002-01-03

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